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Your Vinyl Transfer Workflow (sharing best needledrop practices)*

Discussion in 'Audio Hardware' started by Vocalpoint, May 11, 2011.

  1. john morris

    john morris Everybody's Favorite Quadron

    Location:
    Toronto, Ontario
    Ohh, I could tell this forum some stories about misaligned heads. Oouch! ie We asked for the master of a well known country and Western album. All they had was a copy of a copy. It was the most poor-piss transfering job we had ever heard. Truth: My Unlce used the 1976 commercially released quarter inch 7.5 ips RTR as a master. It was actually better!
     
    Dave112 and ghost rider like this.
  2. Stan94

    Stan94 Forum Resident

    Location:
    Paris, France
    Thanks for your help. Now this is weird: when I plugged my cart the other way around (R is L and L is R), the levels were even. I put it back straight and switched the RCA plugs and same thing, the levels matched (using a true mono record makes it easy to check). But when everything is in place, the left channel is like 2db louder than the right. WTF?
     
  3. Geordiepete

    Geordiepete Tippet tyer

    Location:
    Japan
    I wonder if you could answer this: can this be done with later and current MacBook Pro headphone jacks? I think I remember reading something about Apple removing the mic capability from newer model audio jacks. I have been thinking I'd like to digitize some records, but don't have the cable, usb dongle, or whatnot to link the turntable or amp to the MacBook yet. Someone on this forum told me that I should be able to do it with a Griffen iMic. Also wondering why you didn't bypass the amp. I think Audacity has an RIAA function, so the turntable can feed directly into the software and get processed there.
     
  4. Apesbrain

    Apesbrain Forum Resident

    Location:
    East Coast, USA
    Check your Mac:
    Change the sound input settings on Mac

    BTW, leave Audacity recording level at 100%. Use your source component to set levels.
     
    Geordiepete likes this.
  5. Peacekeepr73

    Peacekeepr73 Digitally Remastered Member

    Location:
    Wyoming, Michigan
    You need a dedicated mic input on the mac, a headphone out port will not do. As for running it though the amp it is necessary to turn the low output signal from the turntable into a line level signal that can be recorded correctly. Audacity may have a RIAA EQ curve but it wont work with a raw turntable output as the signal has not gone through a proper phono preamp.
     
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  6. Stan94

    Stan94 Forum Resident

    Location:
    Paris, France
    May I ask why 37%?
     
  7. JohnO

    JohnO Forum Resident

    Location:
    Washington, DC
    That is 2 dB. Rule of thumb: 3 dB is 50%. But it is not linear from 3 to 2 to 1 dB (it's a dB!) Try 37 for 2 dB but your ears are the final judge.
     
  8. Grant

    Grant Senior Member

    Location:
    United States
    I always have to boost my left channel by about 1 db. but, remember that 3 db is always twice as loud.
     
    arisinwind likes this.
  9. darkmass

    darkmass Forum Resident

    (Emphasis mine.) A little caution is required here. Yes, +3 dB is twice the acoustic power (and it doesn't matter where on the dB scale the 3 dB increase occurs: 83 dB has twice the acoustic power of 80 dB, 15 dB has twice the acoustic power of 12 dB), but "loud" often refers to the human perception of volume levels. When it comes to human perception, a 10 dB increase generally "sounds" twice as loud. References to increase types on this page.

    And as a bridge to a recent discussion in this thread...

    The "Azimuth" function/popup in iZotope Advanced RX versions has gone through a name change, though the functions contained in the popup seem to be identical before and after the name change. Previously the popup was called "Channel Ops" and contained automated and manual separate L/R track adjustments for Channel Balance (under the heading "Level") and for Azimuth (under the heading "Delay"). Currently the popup is called "Azimuth" but the popup still contains the same automated and manual separate L/R track adjustments for Channel Balance and for Azimuth. Personally I felt "Channel Ops" was a better and more meaningful designation for the function/popup, but there you go.


    Within the "Azimuth" function/popup there is a "Suggest" button towards the upper left. Before bringing up the "Azimuth" popup, do an Edit/Select All (or in special circumstances manually select a reduced region of importance in the audio file). The "Azimuth" popup only operates on the selected region. Okay, then once the "Azimuth" popup has been brought up, the "Suggest" button will trigger a mathematical evaluation of channel balance and of channel azimuth. The duration of the Suggest evaluation depends on the playing time the selected region represents and also the sample rate of the file, but it'll probably take a couple of minutes for RX to work up its recommendations to correct channel balance and channel azimuth. RX moves only the right channel sliders; however, the user can move left and right channel sliders any way they want (and that can be done on top of the "Suggest" results). When the user is satisfied with the Balance and Azimuth settings, selecting the "Render" button in the popup will apply the results to the selected region of the file. A little bit of caution is useful for the Balance part of the popup, if the peak level of the audio file is at 0 dB, or near 0 dB, the results of doing the "Suggest" can call for increasing the right channel level, pushing the right channel into saturation. Should it look like this will be happening to a file of mine, within the "Azimuth" popup I'll manually subtract the exact same dB values from the left and right channels before using the "Render" button. This will reduce the overall level of the file a bit, but the channel balance will be maintained and neither channel will be pushed into saturation.

    Now I love the "Azimuth" popup. If I'm doing serious needle dropping, or careful editing for other reasons, I'll do the mathematical RX corrections of channel balance and channel azimuth on every single cut. For a given record side, azimuth can be reasonably consistent for every cut on the record side, but there can still be variations. Maybe think of things this way... Ideally, the exact same equipment and exact same circumstances have been applied to every cut on the side, but that is not guaranteed to be the case. It's a complicated world. Also--and john morris can probably supply some real world clarification--my suspicion is channel balance significantly depends on the talents and objectives of those people in the engineering/mastering chain. What those people do should certainly be respected. But still, I find for headphone listening in particular, a mathematical RX channel balancing can help result in an enveloping and full sound field.
     
    Last edited: Oct 23, 2020
    Grant and stetsonic like this.
  10. clip

    clip Well-Known Member

    Location:
    Australia
    I’ve just read through maybe 50% of the posts and you guys are amazing. So much experience and knowledge. I’m just beginning to transfer some vinyl to digital and this has been enormous help, so thanks! Come a long way from when I used to sit in front of my dads record player and record my albums on my Toshiba cassette recorder using the inbuilt condenser microphone!
     
    Grant likes this.
  11. ghost rider

    ghost rider Forum Resident

    Location:
    Chicago
    l meant -0.3

    I may give it a try thanks for the info. I generally leave the channel mismatch between the tracks as they were mastered. The only time I mess with that is when the whole album or album side is obviously off.
     
    Last edited: Oct 23, 2020
    darkmass likes this.
  12. Stan94

    Stan94 Forum Resident

    Location:
    Paris, France
    Is there a method to build a scale in order to get all the values easily? I suck at maths.
     
  13. Stan94

    Stan94 Forum Resident

    Location:
    Paris, France
    I found the decibel table from March, 1955 and it answered all my questions. Wow.
     
  14. JohnO

    JohnO Forum Resident

    Location:
    Washington, DC
    or +3 dB would be +100%. And -3 dB would be half as loud, or -50%. It is confusing until one works with it and "gets" it.
    March 1955 where?
    There are online calculators too, here's one
    https://www.lasercalculator.com/db-percent-converter/
     
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  15. Grant

    Grant Senior Member

    Location:
    United States
    The older versions of Adobe Audition used to let the user switch between db and percentage in their normalize module. I always preferred to use db.
     
  16. Grant

    Grant Senior Member

    Location:
    United States
    I decided to do a remaster of one of my old needledrops. It sounds very good now, but I rushed though the last part, neglected to configure the limiter, and wound up with some overs. Oh well...
     
  17. ghost rider

    ghost rider Forum Resident

    Location:
    Chicago
    Post a before and after with some explanation as to what has changed. Pick parts that sound good.
     
    arisinwind likes this.
  18. Grant

    Grant Senior Member

    Location:
    United States
    Will do. I just hope I remember to save the old files.
     
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  19. Stan94

    Stan94 Forum Resident

    Location:
    Paris, France
    I make experiments with a free VST compressor/limiter/etc. called No.6
    Last time I used the compressor with +4db, wery little compression like 0.5/1 (the first setting) and the result sounded good. As per Mr. Diament's instructions, the original file is captured below -6db.
     
  20. BendBound

    BendBound Forum Resident

    Location:
    Bend, OR
    Is the -6dB absolute or an average for the recording? I've read that pushing recording equipment (I use a Tascam) towards say 0dB in a dub creates its own issues with the recording. I have tried to get closer to 0dB to avoid applying gain to a recording. Maybe that approach is less than ideal. Is it better, for example, to under record as it were and then process up for a final dub?
     
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  21. anorak2

    anorak2 Forum Resident

    Location:
    Berlin, Germany
    Yes absolutely. Analogue sources have consideably less dynamic range than the digital system, even "mere" 16 bits. So you can leave some safety margin above to be sure you'll avoid clipping, without compromising quality. You can apply gain later to reach exactly 0dB. Riding the gain just below saturation, like is recommendable with analogue tape to minimise hiss, is not necessary with digital.
     
  22. Stan94

    Stan94 Forum Resident

    Location:
    Paris, France
    It is -6db peak. The meters stay in the green zone and gain adjustment is made afterwards. I was worried the volume would be too low but it's not the case.
     
  23. numanoid

    numanoid Forum Resident

    Location:
    Valparaiso, IN
    Crazy, I thought I discovered the “white noise” trick myself and was the only one doing it. The difference for me is that rather than using a white noise generator I use a 12bit dither noise floor for a nice, clean digital hiss that smooths out the random broadband noise of the lp itself. I then sample the new noise profile and remove it all. Sometimes I’ll add more of the 12bit dither as a final step, to even out the tape hiss from the original source. If you want to get fancy, leave your new noise floor between tracks and it sounds like you’re listening to a high quality tape source.

    Adding audible dither also has the benefit of smoothing out any artifacts from declicking and decrackle.

    As for less is more, I tend to disagree. Records impart so many of their own characteristics and flaws to the signal, I feel they should all be removed to the best if your ability. Clicks should not be there and modern removal methods are pretty good, better depending on how much cash you want to throw at the job. I’d rather have the computer interpolate .003 seconds worth of data rather than listen to a click that I know should not be there. All of the sound of the vinyl playing, that low whoosh between tracks, that has to go, as does everything below 20hz. All broadband crackle between tracks that can also be heard in quiet parts and fades, that’s all gotta go too.

    The nice thing is you can do as much or as little as you want and always come back and do more later.
     
  24. BendBound

    BendBound Forum Resident

    Location:
    Bend, OR
    You have introduced a new way to clean up an lp. I don't know how to add a 12bit dither noise floor but if you elaborate I'll try it.

    Obviously, a clean lp is best to record. So I use a 20 minute bath in an 80kHz ultrasonic cleaner to work on my records. Then I take them on a ride with a VPI17 to essentially dry them. All with super clean pads. This procedure is done by VPI owner Harry Weisfeld.

    Then I take an at once recorded 2-sided lp as a 88.2kHz/24bit track into IzotopeRX7. I look for the right-left gain mismatch in recorded volume, and typically add a 1.0dB gain to the left track only. Then I normalize both tracks to 0.3dB. I have not yet tried the routine BrilliantBob suggested for increasing the gain (VST plug in?), I use IzotopeRX7.

    I no longer run ClickRepair as a routine, depending on how noisy the record is. So next, I use the white noise generator routine that BrilliantBob introduced. As you do, I use De-Noise to sample the lead in bit or some other black section of lp, to figure out how to lower the noise level in the entire recording.

    Next I painstakingly I listen to this processed dub to hear clicks, pops and any other vinyl imparted aberration. I remove what I hear.

    Then I parse and name individual tracks, insert a fade out or fade in as the case may be to each track. I save this file as a second master (the first being the original recording).

    Finally, resample the lot (Red Book CDR) and export those individual files, dithered with MBIT+.

    Then I made a CDR, complete with artwork.
     
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  25. Grant

    Grant Senior Member

    Location:
    United States
    I have added low-level white noise to files before to attain the results you speak of, but, how do you apply a file with a different bit-depth to one with another? Are you creating the 12-bit file and then converting it to the bit-depth of the file you're adding it to?
     
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